Make sure you have<br>busydetect=yes<br>busycount=3<br><br>somewhere below your [general] context in chan_dahdi.conf (or zapata.conf depending on your asterisk version) and restart the the service.<br><br>This should be enoough to do the magic.<br>
<br>Alyed<br><br><br><div class="gmail_quote">2010/3/21 Daniel Bareiro <span dir="ltr"><<a href="mailto:daniel-listas@gmx.net">daniel-listas@gmx.net</a>></span><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div class="im">-----BEGIN PGP SIGNED MESSAGE-----<br>
Hash: SHA1<br>
<br>
</div>Hi, Gordon.<br>
<div class="im"><br>
On Sun, 21 Mar 2010, Gordon Henderson wrote:<br>
<br>
>> I'm testing with a Grandstream BT200 telephone and, according to I<br>
>> read, it has a LED that blinks if for that extension messages were<br>
>> left.<br>
>><br>
>> In "Voice Mail UserID", under the "ACCOUNT" tab, I put *100 that is<br>
>> the extension in which my Asterisk answer the voicemail service and<br>
>> if then I press MESSAGE button, the telephone communicates with<br>
>> Asterisk and, after to introduce the password, it indicates to me<br>
>> that I have messages. But the luminous indicator does not work.<br>
>><br>
>> It is necessary to configure something special for this? It can be<br>
>> that it doesn't work because there is to introduce one password<br>
>> previously?<br>
<br>
> There's another setting in the phone you need to set "SUBSCRIBE for<br>
> MWI".<br>
<br>
</div>Yes. I was needing to indicate the use of MWI of the side of the<br>
configuration of the telephone. I selected the "SUBSCRIBES for MWI"<br>
checkbox.<br>
<div class="im"><br>
> And make-sure the mailbox number is listed in the sip.conf entry for<br>
> that phone.<br>
<br>
</div>According to which I was reading, the MWI notifications become by the<br>
option "mailbox=" in the configuration of the extension. For this<br>
extension, the 104, had "mailbox=104" but still with MWI enabled option,<br>
it was not working. After to think enough on this subject, I have<br>
noticed that instead of 104 I had to put 104@voicemail since "voicemail"<br>
it was context that I'm using in voicemail.conf.<br>
<br>
With this already was working.<br>
<br>
However, beyond this, I was with the following situation: if I call from<br>
a cell phone, my Asterisk take the call, it presents to the caller the<br>
possibility to dialing an extension number and, in case of not doing it,<br>
it transfers this call to a specific extension.<br>
<br>
Then, if in this extension nobody takes the call, the service of<br>
voicemail is triggered so that the caller leaves its message from the<br>
cell phone. But if it hangs after to let the message without have<br>
pressed previously the pound key, the channel is taken and no longer any<br>
other call enters the PBX from the PSTN. This does not happen if the<br>
caller presses the pound key after to have left his message.<br>
<br>
As I have a box at which the cable arrives from the PSTN in which there<br>
are two ports of derivation and in one of them it leaves the cable for<br>
the Asterisk PBX (connected only then), after to have detected this<br>
problem I tried connecting in the other port an analog telephone and,<br>
indeed, it did not have tone as if never it had been hung. In addition<br>
this was confirmed because the MWI light never blinked on the telephone.<br>
<br>
After restarting the Asterisk server, yes the MWI light blinks and in<br>
addition I could corob the time in which the channel was "taken" seeing<br>
that the message lasted more than nine minutes.<br>
<br>
To what this problem can be due? It has to do the call is made<br>
specifically from cell phone through the PSTN (because if I leave a<br>
message hanging directly without pressing the pound key from an local<br>
extension, this does not happen)? There is some form to avoid it?<br>
<br>
Thanks for your reply!<br>
<br>
Regards,<br>
Daniel<br>
<br>
-----BEGIN PGP SIGNATURE-----<br>
Version: GnuPG v1.4.9 (GNU/Linux)<br>
<br>
iEYEARECAAYFAkums0oACgkQZpa/GxTmHTcGpQCghJvfphxc5ZzZhouryA+OlwGm<br>
20AAoJP64a2EVeigx08D/5g5XN8oBXgf<br>
=Hskd<br>
-----END PGP SIGNATURE-----<br>
<div><div></div><div class="h5"><br>
<br>
--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</div></div></blockquote></div><br><br clear="all"><br>